[asterisk-bugs] [JIRA] (ASTERISK-20644) Don't always use the existing TCP connection for in-dialog requests

[Olle Johansson (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-20644?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=199221#comment-199221 ] 

Olle Johansson commented on ASTERISK-20644:
-------------------------------------------

If you read the comments I added years ago in chan_sip.c this was listed there. It is also clearly explained in sip.conf.sample:

; Note that the TCP and TLS support for chan_sip is currently considered
; experimental.  Since it is new, all of the related configuration options are
; subject to change in any release.  If they are changed, the changes will
; be reflected in this sample configuration file, as well as in the UPGRADE.txt file.

I would not argue about the RFC with Mr Baz Castillo. He is right here. It will take a lot of code to fix this.
                
> Don't always use the existing TCP connection for in-dialog requests
> -------------------------------------------------------------------
>
>                 Key: ASTERISK-20644
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-20644
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/Interoperability, Channels/chan_sip/TCP-TLS
>    Affects Versions: 11.0.0
>            Reporter: Iñaki Baz Castillo
>
> If Asterisk receives an INVITE via TCP comming from a SIP proxy, answers the call and later Asterisk sends a re-INVITE or BYE for that dialog, Asterisk sends such an in-dialog request over the existing TCP connection previously opened by the SIP proxy. This is incorrect. Asterisk should open a NEW TCP connection to the address given in the top Record-Route header.
> So for example, Asterisk receives the following INVITE from TCP 1.2.3.4 port 8888:
> {code}
> # TCP 1.2.3.4:8888 => ASTERISK:5060
> INVITE sip:test at domain.com SIP/2.0
> Record-Route: <sip:1.2.3.4:5060;transport=tcp>
> {code}
> When Asterisk sends a BYE or a re-INVITE in this leg, it MUST respect the address in the top Route of the BYE or re-INVITE, which is: TCP 1.2.3.4 port 5060. So it should send the BYE to:
> {code}
> # TCP ASTERISK:5060 => 1.2.3.4:5060
> BYE sip:abc at 9.8.7.6 SIP/2.0
> Route: <sip:1.2.3.4:5060;transport=tcp>
> {code}
> This is something really basic in RFC 3261 and mandatory. Asterisk could use the existing TCP connection just in case nat=yes (or some other new values I don't fully know in the latest version), but by default, if "nat" parameter is not set for the Proxy peer, please DON'T reuse the existing TCP connection because that is a violation of RFC 3261.
> In fact, it's perfectly legal for a SIP proxy or SIP server to refuse/reject SIP requests coming from a TCP connection that the proxy/server itself opened against a remote proxy/server. These are RFC 3261 rules, really.

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